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gstreamer/ext/audiofile
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Jan Schmidt 3b47dd30d6 ext/audiofile/gstafsrc.c: Remove old debug output 
Original commit message from CVS:
* ext/audiofile/gstafsrc.c: (gst_afsrc_get):
Remove old debug output
* ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type),
(gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state),
(gst_dvdec_set_property), (gst_dvdec_get_property):
Change the quality setting to an enum, so it works from gst-launch
Don't renegotiate a non-linked pad. Allows audio only decoding.
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps),
(gst_deinterlace_link), (gst_deinterlace_init):
* gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
(gst_videodrop_link):
Some caps negotiation fixes
2004-05-12 14:53:58 +00:00
..
gstaf.c gst-indent 2004-03-14 22:34:33 +00:00
gstafparse.c don't mix tabs and spaces 2004-03-15 19:32:27 +00:00
gstafparse.h *.h: Revert indenting 2004-03-15 16:32:54 +00:00
gstafsink.c don't mix tabs and spaces 2004-03-15 19:32:27 +00:00
gstafsink.h *.h: Revert indenting 2004-03-15 16:32:54 +00:00
gstafsrc.c ext/audiofile/gstafsrc.c: Remove old debug output 2004-05-12 14:53:58 +00:00
gstafsrc.h *.h: Revert indenting 2004-03-15 16:32:54 +00:00
Makefile.am remove audiofile typefinding because it is buggy and we support all of its formats anyway. 2003-11-03 19:18:36 +00:00
README fixed some GST_LIBS stuff added audiofile added gst-libs/audio building 2001-12-21 11:46:15 +00:00

README 

This plugin wraps the SGI Audiofile 
(http://oss.sgi.com/projects/audiofile/) library into a src and sink
element.

You can read from and write to the supported formats (WAVE, AIFF, AIFFC,
NEXTSND).

What is supported :
* all the file formats
* integer sample data, both 2's complement and unsigned
* 8 or 16 bit width & depth (haven't tested others)
* sample rate
* some sort of endianness control

What isn't supported yet :
* float data

What you can do :
* src element only accepts location argument
* sink element accepts location, endianness and type
	- location : file on the system to output
	- endianness : at this time endianness is still a bit shady
	  	you can either set 1234 or 4321;
		setting it to 4321 will byteswap the buffer data
		you might want to keep it at 1234 for now
	- type : one of the file types

Use gstreamer-inspect on afsink and afsrc to see all of the supported
options.

Examples :

* tools/gstreamer-launch afsrc location=/opt/media/wav/dark-480-16-m.wav ! afsink type=2 location=/opt/media/wav/dark-480-16-m.aiff

Future plans :

* add float support
* wrap up afsink and afsrc with pipe and fork to act like data convertors,
  allowing arbitrary choice of sink and src element